In conventional telecommunications systems, the bandwidth employed for transmitting speech signals has typically been limited to a frequency range of about 300 to 3400 Hz with a sampling rate of 8 kHz. This limitation applies to normal PCM (pulse code modulation) speech, which employs a 64 kbit/s coding defined in specification ITU-T G.711, as well as to most of the low-bit rate speech coding methods used in telecommunications systems.
The use of such a narrow frequency rage for speech transmissions reduces naturalness and intelligibility of the speech when presented at the receiving end. Therefore, wideband speech was introduced, which provides a better speech quality to the user of the receiving terminal. Wideband speech codecs (compressor/decompressor) were standardized e.g. in ITU-T G.722, which specifications extend the bandwidth to up to 8 kHz with a sampling frequency of 16 kHz. Current wideband speech transmissions, however, requires bit-transparent end-to-end connections, e.g. by ISDN (Integrated Services Digital Network). Only tandem-free operation (TFO) or transcoder-free operation (TrFO) connections between two wideband terminals allow to fully utilize the wideband terminal capabilities. In addition, it requires special terminals equipped at both transmission ends with the same wideband codecs. These restrictions have limited to date the utilization of wideband speech.
In the future, the importance of wideband speech will increase, as the forthcoming adaptive multirate wideband (AMR-WB) speech codec, standardized in various 3 GPP specifications, will be taken into use for the 2G (second generation) and 3G (third generation) networks. But also AMR-WB will require bit-transparent tandem free operation and AMR-WB capable terminals.
To date, the majority of terminals moreover still uses narrowband speech transmissions, and each connection between a wideband terminal on the one hand and a narrowband terminal on the other hand is narrowband. In order to establish e.g. a call between an AMR-WB capable and a conventional narrowband (NB) terminal, like a PSTN (Public Switched Telephone Network) or a PLMN (Public Land Mobile Network) terminal, either the coding method needs to be negotiated in a way that the AMR-WB terminal shall use a narrow band codec, or AMR-WB speech frames need to be transcoded into narrowband speech and vice versa in the network. In both cases the user of a receiving AMR-WB terminal will experience narrowband speech. Thus there will be an annoying quality difference between AMR-WB to AMR-WB and narrowband to AMR-WB calls. This further reduces the benefits gained with wideband terminals, until such terminals become widely available.
The same problems are encountered also with narrowband services accessed by a user of a wideband terminal, e.g. announcements, voice mail systems, interactive voice interfaces and narrowband audio streaming applications. In case speech-based network services are to be provided for both, wideband and narrowband terminals, the storage capacity required for storing speech samples for both terminal types is moreover tripled compared to the conventional narrowband case.
Another problem with the transmission of wideband audio signals results from the fact that the extended audio bandwidth used by the wideband terminals requires more transmission capacity. More specifically, the transmission capacity requirements are doubled for equal speech coding schemes. In addition, the adoption of wideband speech transmission in wireless communications network is difficult due to the lack of established wideband codecs for telecommunication networks.